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SIP Outbound Call Failing

I am attempting to troubleshoot an outbound call failing across a SIP trunk. I can dial this specific number from outside the network just fine, but it is the only that is failing across the SIP trunk. I have made calls to similar numbers and they are all routing out the same path. When I dial 18667173880, there is silence for about ten seconds, then fast busy, when I dial any other 18XX number, including 18666978487, the call proceeds and connects. I am not very familiar with SIP Debugs, but I see what I think are key messages:

SIP/2.0 408 Request Timeout

SIP/2.0 503 Service Unavailable

Reason: Q.850;cause=102

I do not know how much of the debug logs are neccesarry, but I will paste them here with the failed call (8667173880) as well as the successful call (8666978487)

I don't really know SIP at all, but from forum searching, I've found it could possibly be a codec issue, an h323 gateway on the other end that is not campatible due to the lack of an 'allow' sip h323 command, or possibly that the endpoint is just non-responsive. But why then does it pickup when I call from my cell phone?

Any help would be greatly appreciated.

*Dec  8 21:58:52.893: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:918667173880@XXX.XX.83.253:5060 SIP/2.0

Date: Thu, 08 Dec 2011 20:42:50 GMT

Call-Info: <sip:XXX.XX.232.12:5060>;method="NOTIFY;Event=telephone-event;Duration=500"

Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, PUBLISH

From: "WP Test" <sip:8346@XXX.XX.232.12>;tag=13a3368a-2ecc-4f4d-bdac-c4e79bb85bf6-75292645

Allow-Events: presence, kpml

Supported: 100rel,timer,replaces

Min-SE:  1800

Remote-Party-ID: "WP Test" <sip:8346@XXX.XX.232.12>;party=calling;screen=yes;privacy=off

Content-Length: 0

User-Agent: Cisco-CUCM6.1

To: <sip:918667173880@XXX.XX.83.253>

Contact: <sip:8346@XXX.XX.232.12:5060;transport=tcp>

Expires: 180

Call-ID: 304b4680-ee11214a-1aabfd-ce814ac@XXX.XX.232.12

Via: SIP/2.0/TCP XXX.XX.232.12:5060;branch=z9hG4bK337aa674f39978

CSeq: 101 INVITE

Session-Expires:  86400

Max-Forwards: 70


*Dec  8 21:58:52.901: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:18667173880@XXX.XX.63.17:5318 SIP/2.0

Via: SIP/2.0/UDP XXX.XX.83.253:5060;branch=z9hG4bK12C57B12

Remote-Party-ID: "WP Test" <sip:8346@XXX.XX.83.253>;party=calling;screen=yes;privacy=off

From: "WP Test" <sip:8346@XXX.XX.83.253>;tag=741AC90C-1674

To: <sip:18667173880@XXX.XX.63.17>

Date: Thu, 08 Dec 2011 21:58:52 GMT

Call-ID: A6CD0CBC-211E11E1-BBA881D9-E389BFEC@XXX.XX.83.253

Supported: 100rel,timer,resource-priority,replaces,sdp-anat

Min-SE:  1800

Cisco-Guid: 2798376044-555618785-3147989465-3817455596

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

CSeq: 101 INVITE

Timestamp: 1323381532

Contact: <sip:8346@XXX.XX.83.253:5060>

Expires: 180

Allow-Events: telephone-event

Max-Forwards: 69

Session-Expires:  86400

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 273

v=0

o=CiscoSystemsSIP-GW-UserAgent 4155 9134 IN IP4 XXX.XX.83.253

s=SIP Call

c=IN IP4 XXX.XX.83.253

t=0 0

m=audio 18564 RTP/AVP 18 101

c=IN IP4 XXX.XX.83.253

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20


*Dec  8 21:58:52.901: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying

Via: SIP/2.0/TCP XXX.XX.232.12:5060;branch=z9hG4bK337aa674f39978

From: "WP Test" <sip:8346@XXX.XX.232.12>;tag=13a3368a-2ecc-4f4d-bdac-c4e79bb85bf6-75292645

To: <sip:918667173880@XXX.XX.83.253>

Date: Thu, 08 Dec 2011 21:58:52 GMT

Call-ID: 304b4680-ee11214a-1aabfd-ce814ac@XXX.XX.232.12

CSeq: 101 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0


*Dec  8 21:58:52.969: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying

Via: SIP/2.0/UDP XXX.XX.83.253:5060;branch=z9hG4bK12C57B12

From: "WP Test" <sip:8346@XXX.XX.83.253>;tag=741AC90C-1674

To: <sip:18667173880@XXX.XX.63.17>

Call-ID: A6CD0CBC-211E11E1-BBA881D9-E389BFEC@XXX.XX.83.253

CSeq: 101 INVITE

Timestamp: 1323381532


*Dec  8 21:59:01.481: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 408 Request Timeout

Via: SIP/2.0/UDP XXX.XX.83.253:5060;branch=z9hG4bK12C57B12

From: "WP Test" <sip:8346@XXX.XX.83.253>;tag=741AC90C-1674

To: <sip:18667173880@XXX.XX.63.17>;tag=aprqngfrt-384igp00000a6

Call-ID: A6CD0CBC-211E11E1-BBA881D9-E389BFEC@XXX.XX.83.253

CSeq: 101 INVITE

Timestamp: 1323381532

Content-Length: 0


*Dec  8 21:59:01.485: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 503 Service Unavailable

Via: SIP/2.0/TCP XXX.XX.232.12:5060;branch=z9hG4bK337aa674f39978

From: "WP Test" <sip:8346@XXX.XX.232.12>;tag=13a3368a-2ecc-4f4d-bdac-c4e79bb85bf6-75292645

To: <sip:918667173880@XXX.XX.83.253>;tag=741AEA90-9E8

Date: Thu, 08 Dec 2011 21:58:52 GMT

Call-ID: 304b4680-ee11214a-1aabfd-ce814ac@XXX.XX.232.12

CSeq: 101 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Reason: Q.850;cause=102

Content-Length: 0

_________________________________________________________________________________________

293764: *Dec  8 20:43:38.693: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:918666978487@XXX.XX.83.254:5060 SIP/2.0

Date: Thu, 08 Dec 2011 20:42:15 GMT

Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, PUBLISH

From: "WP Test" <sip:8043258346@XXX.XX.232.12>;tag=13a3368a-2ecc-4f4d-bdac-c4e79bb85bf6-75292555

Allow-Events: presence

Supported: 100rel,timer,replaces

Min-SE:  1800

Remote-Party-ID: "WP Test" <sip:8043258346@XXX.XX.232.12>;party=calling;screen=yes;privacy=off

Content-Length: 0

User-Agent: Cisco-CUCM6.1

To: <sip:918666978487@XXX.XX.83.254>

Contact: <sip:8043258346@XXX.XX.232.12:5060;transport=tcp>

Expires: 180

Call-ID: 1b6eb300-ee112127-1aabee-ce814ac@XXX.XX.232.12

Via: SIP/2.0/TCP XXX.XX.232.12:5060;branch=z9hG4bK
337a6f4935e974

CSeq: 101 INVITE

Session-Expires:  86400

Max-Forwards: 70


293765: *Dec  8 20:43:38.705: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying

Via: SIP/2.0/TCP XXX.XX.232.12:5060;branch=z9hG4bK337a6f4935e974

From: "WP Test" <sip:8043258346@XXX.XX.232.12>;tag=13a3368a-2ecc-4f4d-bdac-c4e79bb85bf6-75292555

To: <sip:918666978487@XXX.XX.83.254>

Date: Thu, 08 Dec 2011 20:43:38 GMT

Call-ID: 1b6eb300-ee112127-1aabee-ce814ac@XXX.XX.232.12

CSeq: 101 INVITE

Allow-Events: t
elephone-event

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

 


293767: *Dec  8 20:43:40.801: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
PRACK sip:918666978487@XXX.XX.83.254:5060;transport=tcp SIP/2.0

Date: Thu, 08 Dec 2011 20:42:15 GMT

From: "WP Test" <sip:8043258346@XXX.XX.232.12>;tag=13a3368a-2ecc-4f4d-bdac-c4e79bb85bf6-75292555

RAck: 596 101 INVITE

Allow-Events: presence

Content-Length: 236

To: <sip:918666978487@XXX.XX.83.254>;tag=591370B8-D02

C
ontent-Type: application/sdp

Call-ID: 1b6eb300-ee112127-1aabee-ce814ac@XXX.XX.232.12

Via: SIP/2.0/TCP XXX.XX.232.12:5060;branch=z9hG4bK337a713ddf3482

CSeq: 102 PRACK

Max-Forwards: 70

v=0

o=CiscoSystemsCCM-SIP 2000 2 IN IP4 XXX.XX.232.12

s=SIP Call

c=IN IP4 XXX.XX.83.45

t=0 0

m=audio 16964 RTP/AVP 18 101

a=rtpmap:18 G729/8000

a=ptime:20

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

I hope someone can assist. Thank you for your time.

Everyone's tags (7)
3 REPLIES

SIP Outbound Call Failing

Hi.

From your attached log, first thing I can see is that first call goes to XXX.XX.83.253 and receives a timeout

the second call goes to XXX.XX.83.254 and it succesfully connects.

Try to check if there is any problem with .253 sip connection.

You can attach vg config here so we can help you more.

HTH

Regards

Carlo

Please rate all helpful posts "The more you help the more you learn"
New Member

SIP Outbound Call Failing

Carlo,

The .253 is one of the SIP trunks in the route list that is offline. I have taken that trunk out of the list and the problem still exists. Apologies, I should have captured the SIP debug when that trunk was not in the list.

version 12.4

service timestamps debug datetime msec

service timestamps log datetime msec

service password-encryption

service sequence-numbers

!

hostname SIP3845-01

!

boot-start-marker

boot-end-marker

!

logging message-counter syslog

logging buffered 5000000

no logging console

!

no aaa new-model

!

dot11 syslog

no ip source-route

ip cef

!

!

!

!

ip name-server XXX.XX.62.36

no ipv6 cef

!

multilink bundle-name authenticated

!

!

!

!

!

voice-card 0

dspfarm

dsp services dspfarm

!

!

!

voice service voip

address-hiding

allow-connections sip to sip

fax protocol cisco

sip

  bind control source-interface Loopback0

  bind media source-interface Loopback0

  early-offer forced

  midcall-signaling passthru

!

!

!

voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g729r8

!

!

!

!

!

voice class sip-profiles 1

request INVITE sip-header Diversion modify "<>" "<>8043258700@XXX.XX.83.254"

request INVITE sip-header Diversion add "Diversion: \"Company\" <>8043258700@XXX.XX.83.254>;reason=unconditional;privacy=off;screen=yes"

!

!

!

!

!

!

!

!

!

!

voice translation-rule 9

rule 1 /^9\(.*\)/ /\1/

!

voice translation-rule 110

rule 1 /^804281....\(.*\)/ /\1/

!

!

voice translation-profile DIGITSTRIP-9

translate called 9

!

!

!

!

!

!

archive

log config

  hidekeys

!

!

!

!

!

ip ftp source-interface Loopback0

!

!

!

!

interface Loopback0

ip address XXX.XX.83.254 255.255.255.255

!

interface GigabitEthernet0/0

description Cisco UBE inside interface

ip address XXX.XX.64.150 255.255.255.252

duplex auto

speed auto

media-type rj45

!

interface GigabitEthernet0/1

description connection to VzB Private IP network

ip address XXX.XX.64.154 255.255.255.252

shutdown

duplex auto

speed auto

media-type rj45

!

router eigrp 2784

network XXX.XX.0.0

network XXX.XX.0.0

no auto-summary

!

ip forward-protocol nd

ip route 0.0.0.0 0.0.0.0 XXX.XX.64.149

ip http server

ip http access-class 23

ip http authentication local

no ip http secure-server

ip http timeout-policy idle 60 life 86400 requests 10000

!

!

!

!

!

!

!

!

!

!

control-plane

!

!

!

call filter match-list 1 voice

outgoing called-number 9558252

!

!

!

sccp local Loopback0

sccp ccm XXX.XX.232.4 identifier 3 priority 3 version 6.0

sccp ccm XXX.XX.232.20 identifier 2 priority 2 version 6.0

sccp ccm XXX.XX.232.12 identifier 1 priority 1 version 6.0

sccp

!

sccp ccm group 10

associate ccm 1 priority 1

associate ccm 2 priority 2

associate ccm 3 priority 3

associate profile 2 register SIP01-MTP

!

dspfarm profile 2 mtp

codec g729r8

maximum sessions software 50

associate application SCCP

!

!

dial-peer voice 100 voip

description OUTBOUND Voice SIP calls to VzB

translation-profile outgoing DIGITSTRIP-9

destination-pattern 9T

voice-class sip early-offer forced

voice-class sip profiles 1

session protocol sipv2

session target sip-server

dtmf-relay rtp-nte

ip qos dscp af32 signaling

no vad

!

dial-peer voice 101 voip

description INBOUND G729 Voice SIP calls from VzB

shutdown

destination-pattern .T

voice-class sip early-offer forced

voice-class sip profiles 1

session protocol sipv2

session target sip-server

incoming called-number .

dtmf-relay rtp-nte

ip qos dscp af32 signaling

no vad

!

dial-peer voice 102 voip

description INBOUND G729 Voice SIP calls from VzB

session protocol sipv2

session target sip-server

incoming called-number [2-9].........$

dtmf-relay rtp-nte

no vad

!

dial-peer voice 200 voip

description To/From CCM02

preference 1

destination-pattern 804.......$

session protocol sipv2

session target ipv4:XXX.XX.232.12

dtmf-relay rtp-nte

fax rate disable

no vad

!

dial-peer voice 201 voip

description To/From CCM03

preference 2

destination-pattern 804.......$

session protocol sipv2

session target ipv4:XXX.XX.232.20

dtmf-relay rtp-nte

fax rate disable

no vad

!

dial-peer voice 202 voip

description To/From CCM01

preference 3

shutdown

destination-pattern 804.......$

session protocol sipv2

session target ipv4:XXX.XX.232.4

dtmf-relay rtp-nte

no vad

!

dial-peer voice 210 voip

description To/From CCM01

preference 1

shutdown

destination-pattern 8043258715$

session protocol sipv2

session target ipv4:XXX.XX.232.12

dtmf-relay rtp-nte

codec g711ulaw

no vad

!

dial-peer voice 9911 voip

translation-profile outgoing DIGITSTRIP-9

destination-pattern 9911

voice-class sip asserted-id pai

session protocol sipv2

session target sip-server

dtmf-relay rtp-nte

ip qos dscp af32 signaling

no vad

!

dial-peer voice 911 voip

destination-pattern 911

voice-class sip asserted-id pai

no voice-class sip early-offer forced

session protocol sipv2

session target sip-server

dtmf-relay rtp-nte

ip qos dscp af32 signaling

no vad

!

!

sip-ua

set pstn-cause 1 sip-status 503

set pstn-cause 102 sip-status 503

retry invite 2

retry bye 2

retry cancel 2

timers trying 1000

sip-server ipv4:XXX.XX.62.49:5389

g729-annexb override

!

!

New Member

SIP Outbound Call Failing

Hello Shawnangelo,

did you get and answer to your issue?

Im having the same problem.

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